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/*
* Copyright (C) 2003-2014 The Music Player Daemon Project
* http://www.musicpd.org
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "config.h"
#include "SlesOutputPlugin.hxx"
#include "Object.hxx"
#include "Engine.hxx"
#include "Play.hxx"
#include "AndroidSimpleBufferQueue.hxx"
#include "../../OutputAPI.hxx"
#include "util/Macros.hxx"
#include "util/Error.hxx"
#include "util/Domain.hxx"
#include "system/ByteOrder.hxx"
#include "Log.hxx"
#include <SLES/OpenSLES.h>
#include <SLES/OpenSLES_Android.h>
class SlesOutput {
static constexpr unsigned N_BUFFERS = 3;
static constexpr size_t BUFFER_SIZE = 65536;
AudioOutput base;
SLES::Object engine_object, mix_object, play_object;
SLES::Play play;
SLES::AndroidSimpleBufferQueue queue;
/**
* This mutex protects the attributes "next" and "filled". It
* is only needed while playback is launched, when the initial
* buffers are being enqueued in the caller thread, while
* another thread may invoke the registered callback.
*/
Mutex mutex;
Cond cond;
bool pause, cancel;
/**
* The number of buffers queued to OpenSLES.
*/
unsigned n_queued;
/**
* The index of the next buffer to be enqueued.
*/
unsigned next;
/**
* Does the "next" buffer already contain synthesised samples?
* This can happen when PCMSynthesiser::Synthesise() has been
* called, but the OpenSL/ES buffer queue was full. The
* buffer will then be postponed.
*/
unsigned filled;
/**
* An array of buffers. It's one more than being managed by
* OpenSL/ES, and the one not enqueued (see attribute #next)
* will be written to.
*/
uint8_t buffers[N_BUFFERS][BUFFER_SIZE];
public:
SlesOutput()
:base(sles_output_plugin) {}
operator AudioOutput *() {
return &base;
}
bool Initialize(const config_param ¶m, Error &error) {
return base.Configure(param, error);
}
bool Configure(const config_param ¶m, Error &error);
bool Open(AudioFormat &audio_format, Error &error);
void Close();
unsigned Delay() {
return pause && !cancel ? 100 : 0;
}
size_t Play(const void *chunk, size_t size, Error &error);
void Drain();
void Cancel();
bool Pause();
private:
void PlayedCallback();
/**
* OpenSL/ES callback which gets invoked when a buffer has
* been consumed. It synthesises and enqueues the next
* buffer.
*/
static void PlayedCallback(gcc_unused SLAndroidSimpleBufferQueueItf caller,
void *pContext)
{
SlesOutput &sles = *(SlesOutput *)pContext;
sles.PlayedCallback();
}
};
static constexpr Domain sles_domain("sles");
inline bool
SlesOutput::Configure(const config_param &, Error &)
{
return true;
}
inline bool
SlesOutput::Open(AudioFormat &audio_format, Error &error)
{
SLresult result;
SLObjectItf _object;
result = slCreateEngine(&_object, 0, nullptr, 0,
nullptr, nullptr);
if (result != SL_RESULT_SUCCESS) {
error.Set(sles_domain, int(result), "slCreateEngine() failed");
return false;
}
engine_object = SLES::Object(_object);
result = engine_object.Realize(false);
if (result != SL_RESULT_SUCCESS) {
error.Set(sles_domain, int(result), "Engine.Realize() failed");
engine_object.Destroy();
return false;
}
SLEngineItf _engine;
result = engine_object.GetInterface(SL_IID_ENGINE, &_engine);
if (result != SL_RESULT_SUCCESS) {
error.Set(sles_domain, int(result),
"Engine.GetInterface(IID_ENGINE) failed");
engine_object.Destroy();
return false;
}
SLES::Engine engine(_engine);
result = engine.CreateOutputMix(&_object, 0, nullptr, nullptr);
if (result != SL_RESULT_SUCCESS) {
error.Set(sles_domain, int(result),
"Engine.CreateOutputMix() failed");
engine_object.Destroy();
return false;
}
mix_object = SLES::Object(_object);
result = mix_object.Realize(false);
if (result != SL_RESULT_SUCCESS) {
error.Set(sles_domain, int(result),
"Mix.Realize() failed");
mix_object.Destroy();
engine_object.Destroy();
return false;
}
SLDataLocator_AndroidSimpleBufferQueue loc_bufq = {
SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,
N_BUFFERS,
};
SLDataFormat_PCM format_pcm;
format_pcm.formatType = SL_DATAFORMAT_PCM;
format_pcm.numChannels = 1;
/* from the Android NDK docs: "Note that the field samplesPerSec is
actually in units of milliHz, despite the misleading name." */
format_pcm.samplesPerSec = audio_format.sample_rate * 1000u;
format_pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
format_pcm.containerSize = SL_PCMSAMPLEFORMAT_FIXED_16;
format_pcm.channelMask = SL_SPEAKER_FRONT_CENTER;
format_pcm.endianness = IsLittleEndian()
? SL_BYTEORDER_LITTLEENDIAN
: SL_BYTEORDER_BIGENDIAN;
SLDataSource audioSrc = { &loc_bufq, &format_pcm };
SLDataLocator_OutputMix loc_outmix = {
SL_DATALOCATOR_OUTPUTMIX,
mix_object,
};
SLDataSink audioSnk = {
&loc_outmix,
nullptr,
};
const SLInterfaceID ids2[] = {
SL_IID_PLAY,
SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
SL_IID_ANDROIDCONFIGURATION,
};
static constexpr SLboolean req2[] = {
SL_BOOLEAN_TRUE,
SL_BOOLEAN_TRUE,
SL_BOOLEAN_TRUE,
};
result = engine.CreateAudioPlayer(&_object, &audioSrc, &audioSnk,
ARRAY_SIZE(ids2), ids2, req2);
if (result != SL_RESULT_SUCCESS) {
error.Set(sles_domain, int(result),
"Engine.CreateAudioPlayer() failed");
mix_object.Destroy();
engine_object.Destroy();
return false;
}
play_object = SLES::Object(_object);
SLAndroidConfigurationItf android_config;
if (play_object.GetInterface(SL_IID_ANDROIDCONFIGURATION,
&android_config) == SL_RESULT_SUCCESS) {
SLint32 stream_type = SL_ANDROID_STREAM_MEDIA;
(*android_config)->SetConfiguration(android_config,
SL_ANDROID_KEY_STREAM_TYPE,
&stream_type,
sizeof(stream_type));
}
result = play_object.Realize(false);
if (result != SL_RESULT_SUCCESS) {
error.Set(sles_domain, int(result),
"Play.Realize() failed");
play_object.Destroy();
mix_object.Destroy();
engine_object.Destroy();
return false;
}
SLPlayItf _play;
result = play_object.GetInterface(SL_IID_PLAY, &_play);
if (result != SL_RESULT_SUCCESS) {
error.Set(sles_domain, int(result),
"Play.GetInterface(IID_PLAY) failed");
play_object.Destroy();
mix_object.Destroy();
engine_object.Destroy();
return false;
}
play = SLES::Play(_play);
SLAndroidSimpleBufferQueueItf _queue;
result = play_object.GetInterface(SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
&_queue);
if (result != SL_RESULT_SUCCESS) {
error.Set(sles_domain, int(result),
"Play.GetInterface(IID_ANDROIDSIMPLEBUFFERQUEUE) failed");
play_object.Destroy();
mix_object.Destroy();
engine_object.Destroy();
return false;
}
queue = SLES::AndroidSimpleBufferQueue(_queue);
result = queue.RegisterCallback(PlayedCallback, (void *)this);
if (result != SL_RESULT_SUCCESS) {
error.Set(sles_domain, int(result),
"Play.RegisterCallback() failed");
play_object.Destroy();
mix_object.Destroy();
engine_object.Destroy();
return false;
}
result = play.SetPlayState(SL_PLAYSTATE_PLAYING);
if (result != SL_RESULT_SUCCESS) {
error.Set(sles_domain, int(result),
"Play.SetPlayState(PLAYING) failed");
play_object.Destroy();
mix_object.Destroy();
engine_object.Destroy();
return false;
}
pause = cancel = false;
n_queued = 0;
next = 0;
filled = 0;
// TODO: support other sample formats
audio_format.format = SampleFormat::S16;
// TODO: support stereo
audio_format.channels = 1;
return true;
}
inline void
SlesOutput::Close()
{
play.SetPlayState(SL_PLAYSTATE_STOPPED);
play_object.Destroy();
mix_object.Destroy();
engine_object.Destroy();
}
inline size_t
SlesOutput::Play(const void *chunk, size_t size, Error &error)
{
cancel = false;
if (pause) {
SLresult result = play.SetPlayState(SL_PLAYSTATE_PLAYING);
if (result != SL_RESULT_SUCCESS) {
error.Set(sles_domain, int(result),
"Play.SetPlayState(PLAYING) failed");
return false;
}
pause = false;
}
const ScopeLock protect(mutex);
assert(filled < BUFFER_SIZE);
while (n_queued == N_BUFFERS) {
assert(filled == 0);
cond.wait(mutex);
}
size_t nbytes = std::min(BUFFER_SIZE - filled, size);
memcpy(buffers[next] + filled, chunk, nbytes);
filled += nbytes;
if (filled < BUFFER_SIZE)
return nbytes;
SLresult result = queue.Enqueue(buffers[next], BUFFER_SIZE);
if (result != SL_RESULT_SUCCESS) {
error.Set(sles_domain, int(result),
"AndroidSimpleBufferQueue.Enqueue() failed");
return 0;
}
++n_queued;
next = (next + 1) % N_BUFFERS;
filled = 0;
return nbytes;
}
inline void
SlesOutput::Drain()
{
const ScopeLock protect(mutex);
assert(filled < BUFFER_SIZE);
while (n_queued > 0)
cond.wait(mutex);
}
inline void
SlesOutput::Cancel()
{
pause = true;
cancel = true;
SLresult result = play.SetPlayState(SL_PLAYSTATE_PAUSED);
if (result != SL_RESULT_SUCCESS)
FormatError(sles_domain, "Play.SetPlayState(PAUSED) failed");
result = queue.Clear();
if (result != SL_RESULT_SUCCESS)
FormatWarning(sles_domain,
"AndroidSimpleBufferQueue.Clear() failed");
const ScopeLock protect(mutex);
n_queued = 0;
filled = 0;
}
inline bool
SlesOutput::Pause()
{
cancel = false;
if (pause)
return true;
pause = true;
SLresult result = play.SetPlayState(SL_PLAYSTATE_PAUSED);
if (result != SL_RESULT_SUCCESS) {
FormatError(sles_domain, "Play.SetPlayState(PAUSED) failed");
return false;
}
return true;
}
inline void
SlesOutput::PlayedCallback()
{
const ScopeLock protect(mutex);
assert(n_queued > 0);
--n_queued;
cond.signal();
}
static bool
sles_test_default_device()
{
/* this is the default output plugin on Android, and it should
be available in any case */
return true;
}
static AudioOutput *
sles_output_init(const config_param ¶m, Error &error)
{
SlesOutput *sles = new SlesOutput();
if (!sles->Initialize(param, error) ||
!sles->Configure(param, error)) {
delete sles;
return nullptr;
}
return *sles;
}
static void
sles_output_finish(AudioOutput *ao)
{
SlesOutput *sles = (SlesOutput *)ao;
delete sles;
}
static bool
sles_output_open(AudioOutput *ao, AudioFormat &audio_format, Error &error)
{
SlesOutput &sles = *(SlesOutput *)ao;
return sles.Open(audio_format, error);
}
static void
sles_output_close(AudioOutput *ao)
{
SlesOutput &sles = *(SlesOutput *)ao;
sles.Close();
}
static unsigned
sles_output_delay(AudioOutput *ao)
{
SlesOutput &sles = *(SlesOutput *)ao;
return sles.Delay();
}
static size_t
sles_output_play(AudioOutput *ao, const void *chunk, size_t size,
Error &error)
{
SlesOutput &sles = *(SlesOutput *)ao;
return sles.Play(chunk, size, error);
}
static void
sles_output_drain(AudioOutput *ao)
{
SlesOutput &sles = *(SlesOutput *)ao;
sles.Drain();
}
static void
sles_output_cancel(AudioOutput *ao)
{
SlesOutput &sles = *(SlesOutput *)ao;
sles.Cancel();
}
static bool
sles_output_pause(AudioOutput *ao)
{
SlesOutput &sles = *(SlesOutput *)ao;
return sles.Pause();
}
const struct AudioOutputPlugin sles_output_plugin = {
"sles",
sles_test_default_device,
sles_output_init,
sles_output_finish,
nullptr,
nullptr,
sles_output_open,
sles_output_close,
sles_output_delay,
nullptr,
sles_output_play,
sles_output_drain,
sles_output_cancel,
sles_output_pause,
nullptr,
};
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